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avcodec/opusdec: switch to swresample
This also fixes linking failures in doc/examples which where apparently caused by the linking order between avcodec and avresample Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
parent
96cb4c8718
commit
ffa05e0802
4
configure
vendored
4
configure
vendored
@ -2110,7 +2110,7 @@ nellymoser_decoder_select="mdct sinewin"
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nellymoser_encoder_select="audio_frame_queue mdct sinewin"
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nuv_decoder_select="dsputil lzo"
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on2avc_decoder_select="mdct"
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opus_decoder_deps="avresample"
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opus_decoder_deps="swresample"
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png_decoder_select="zlib"
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png_encoder_select="dsputil zlib"
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prores_decoder_select="dsputil"
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@ -5140,7 +5140,7 @@ enabled subtitles_filter && prepend avfilter_deps "avformat avcodec"
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enabled lavfi_indev && prepend avdevice_deps "avfilter"
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enabled opus_decoder && prepend avcodec_deps "avresample"
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enabled opus_decoder && prepend avcodec_deps "swresample"
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expand_deps(){
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lib_deps=${1}_deps
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@ -29,7 +29,7 @@
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#include "libavutil/float_dsp.h"
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#include "libavutil/frame.h"
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#include "libavresample/avresample.h"
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#include "libswresample/swresample.h"
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#include "avcodec.h"
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#include "get_bits.h"
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@ -137,7 +137,7 @@ typedef struct OpusStreamContext {
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float *out_dummy;
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int out_dummy_allocated_size;
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AVAudioResampleContext *avr;
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SwrContext *swr;
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AVAudioFifo *celt_delay;
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int silk_samplerate;
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/* number of samples we still want to get from the resampler */
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@ -40,7 +40,7 @@
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "libavresample/avresample.h"
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#include "libswresample/swresample.h"
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#include "avcodec.h"
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#include "celp_filters.h"
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@ -114,9 +114,9 @@ static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
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{
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int celt_size = av_audio_fifo_size(s->celt_delay);
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int ret, i;
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ret = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size, nb_samples,
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NULL, 0, 0);
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ret = swr_convert(s->swr,
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(uint8_t**)s->out, nb_samples,
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NULL, 0);
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if (ret < 0)
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return ret;
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else if (ret != nb_samples) {
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@ -159,15 +159,16 @@ static int opus_init_resample(OpusStreamContext *s)
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uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
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int ret;
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av_opt_set_int(s->avr, "in_sample_rate", s->silk_samplerate, 0);
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ret = avresample_open(s->avr);
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av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0);
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ret = swr_init(s->swr);
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if (ret < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
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return ret;
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}
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ret = avresample_convert(s->avr, NULL, 0, 0, delayptr, sizeof(delay),
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silk_resample_delay[s->packet.bandwidth]);
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ret = swr_convert(s->swr,
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NULL, 0,
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delayptr, silk_resample_delay[s->packet.bandwidth]);
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if (ret < 0) {
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av_log(s->avctx, AV_LOG_ERROR,
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"Error feeding initial silence to the resampler.\n");
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@ -218,7 +219,7 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
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/* decode the silk frame */
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if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
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if (!avresample_is_open(s->avr)) {
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if (!swr_is_initialized(s->swr)) {
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ret = opus_init_resample(s);
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if (ret < 0)
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return ret;
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@ -232,12 +233,9 @@ static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size
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av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
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return samples;
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}
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samples = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size,
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s->packet.frame_duration,
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(uint8_t**)s->silk_output,
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sizeof(s->silk_buf[0]),
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samples);
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samples = swr_convert(s->swr,
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(uint8_t**)s->out, s->packet.frame_duration,
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(uint8_t**)s->silk_output, samples);
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if (samples < 0) {
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av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
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return samples;
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@ -374,10 +372,10 @@ static int opus_decode_subpacket(OpusStreamContext *s,
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int i, j, ret;
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/* check if we need to flush the resampler */
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if (avresample_is_open(s->avr)) {
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if (swr_is_initialized(s->swr)) {
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if (buf) {
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int64_t cur_samplerate;
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av_opt_get_int(s->avr, "in_sample_rate", 0, &cur_samplerate);
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av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
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flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
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} else {
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flush_needed = !!s->delayed_samples;
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@ -406,7 +404,7 @@ static int opus_decode_subpacket(OpusStreamContext *s,
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av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
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return ret;
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}
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avresample_close(s->avr);
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swr_close(s->swr);
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output_samples += s->delayed_samples;
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s->delayed_samples = 0;
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@ -555,7 +553,7 @@ static av_cold void opus_decode_flush(AVCodecContext *ctx)
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if (s->celt_delay)
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av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
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avresample_close(s->avr);
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swr_close(s->swr);
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ff_silk_flush(s->silk);
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ff_celt_flush(s->celt);
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@ -577,7 +575,7 @@ static av_cold int opus_decode_close(AVCodecContext *avctx)
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s->out_dummy_allocated_size = 0;
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av_audio_fifo_free(s->celt_delay);
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avresample_free(&s->avr);
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swr_free(&s->swr);
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}
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av_freep(&c->streams);
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@ -627,16 +625,17 @@ static av_cold int opus_decode_init(AVCodecContext *avctx)
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s->fdsp = &c->fdsp;
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s->avr = avresample_alloc_context();
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if (!s->avr)
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s->swr =swr_alloc();
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if (!s->swr)
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goto fail;
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layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
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av_opt_set_int(s->avr, "in_sample_fmt", avctx->sample_fmt, 0);
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av_opt_set_int(s->avr, "out_sample_fmt", avctx->sample_fmt, 0);
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av_opt_set_int(s->avr, "in_channel_layout", layout, 0);
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av_opt_set_int(s->avr, "out_channel_layout", layout, 0);
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av_opt_set_int(s->avr, "out_sample_rate", avctx->sample_rate, 0);
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av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0);
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av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0);
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av_opt_set_int(s->swr, "in_channel_layout", layout, 0);
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av_opt_set_int(s->swr, "out_channel_layout", layout, 0);
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av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0);
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av_opt_set_int(s->swr, "filter_size", 16, 0);
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ret = ff_silk_init(avctx, &s->silk, s->output_channels);
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if (ret < 0)
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